In present-day sound pick-up systems, acoustic antennae are formed by several microphones, which are configured in such a way as to carry out sound pick-up aiming to cause only slight degradation of the sound signal. To this end, a programmable pure delay signal is introduced into the signal delivered by each microphone, so as either to correct for the physical geometry of the acoustic antenna produced, or to steer the receiving pattern of the acoustic antenna in a given direction with respect to the axis of symmetry of the acoustic antenna. The signals appropriately delayed for each microphone are then summed in order to deliver a resultant signal representative of the sound signal.
A first solution for performing this delay function consists in inserting analog delay lines in cascade with each microphone. This solution, however, exhibits the drawbacks and limitations of analog systems, that is to say inadequate reproducibility and precision for professional applications.
A second solution may consist, however, on the basis of a signal digitized either after each microphone or after summing, this digitized signal consisting of successive samples separated by a sampling period T, in storing these samples in memory. This is because the memory storage of N successive samples has the effect of delaying the transmission of the signal, subjected to this memory storage, by a delay duration of .DELTA.=N.times.T. The delay .DELTA. must however be greater than the duration of the sampling period T, with the desired precision. For sampling frequencies of 16 kHz and 48 kHz, the sampling period is equal to 62.5 .mu.s and 20.8 .mu.s respectively.
In certain specific professional applications, the values of sampling period are greater, with a precision of 1 .mu.s.
In such a case, it is not possible to apply the delay directly to the samples arising from the analog/digital conversion.
It is then necessary to raise the frequency of sampling of the signal delivered by the microphones, by oversampling. For a frequency of the samples, after oversampling, equal to F.sub.2, the frequency of the samples before oversampling being equal to F.sub.1, the duration of the period of the samples after oversampling is divided in the ratio K=F.sub.2 /F.sub.1. The value of this ratio is determined by the value of F.sub.1, the sampling frequency, and by the precision sought for the delay applied.
For example, for F.sub.1 =16 kHz and for a precision, the minimum delay, .DELTA.m=1 .mu.s, the value of K is K=63. The value adopted is the value, the immediately higher multiple of two, K=64.
Such a device is represented diagrammatically in FIG. 1a in which T.sub.2 =1/F.sub.2 is much less than 1 .mu.s, the delay applied, .DELTA.=M.times.T.sub.2, corresponding to the memory storage of M samples. Passing the delayed oversampled signal through a sub-sampling filter, of ratio 1/K, performing the inverse operation, brings the delayed signal back to the sampling frequency F.sub.1.
Such a device, described particularly in the document U.S. Pat. No. 3,997,772, makes it possible to introduce a programmed delay of less than the sampling period. However, no integrated component exists capable of performing such a function.
In a general way, as represented in FIG. 1b, it is necessary to use a digital signal processor DSP to implement the oversampling filter, management of the samples in the external memory and the sub-sampling filter. The delay is obtained by writing the samples into the external memory and rereading previously memory-stored samples. The choice of the over- and sub-sampling filters is important. If they are chosen with many coefficients, frequency response is perfect, at the price of high computing power (about 4 MIPS--Mega Instructions Per Second--for F.sub.1 =16 kHz and filters with 256 coefficients), although out-of-band rejection of 70 dB nevertheless causes a slight degradation of the signal.
If, on the other hand, they are chosen to have a lower number of coefficients, 64 coefficients for F.sub.1 =16 kHz, the computing power is certainly reduced to 1 MIPS, but the response curve is greatly degraded, by the presence of undulations in the passband and by poor out-of-band rejection, less than 30 dB, entailing a significant degradation in the signal.
Hence, for a small acoustic antenna including 10 microphones, implementing such delay circuits requires computing power lying between 10 and 40 MIPS. For an acoustic antenna including 64 microphones, an antenna more particularly intended for video conferencing applications, the computing power necessary lies between 64 and 256 MIPS, which requires the use of 2 to 6 DSPs, digital signal processors, to which must be added the external delay memories. The cost of implementing such antennae, even in the case of small antennae, is therefore very high.
In current techniques for analog/digital conversion, a conventional converter usually comprises an anti-aliasing analog filter, in accordance with the Shannon theorem, the signal obtained, limited in frequency, being converted by successive approximation with respect to a reference voltage. The precision of the converter depends on the precision of the comparison with the reference voltage. This type of device requires an analog filter, the cutoff frequency of which is a function of the sampling frequency, which makes it necessary to change the analog filter when changing the sampling frequency. These converters are therefore very inflexible in use.
More recently, another category of analog/digital converters has been developed, these converters including a plurality of conversion stages in cascade each driven by a different sampling frequency.
Such a type of converter is represented in FIG. 1c, and corresponds in a non-limiting way to a converter known as a "delta sigma" converter, denoted .DELTA..SIGMA. converter.
Following first-order low-pass analog filtering, a converter stage quantizes the filtered analog signal at a high rate, F.sub.1 =4 MHz (4096 kHz) for a sampling frequency of F.sub.3 =16 kHz, over a low number of bits, 1, 2 or 3 bits, by virtue of a .DELTA..SIGMA. conversion module shifting the quantization noise to the high frequencies. The converted signal is brought back to an intermediate frequency, for example F.sub.1 =16.times.F.sub.2, i.e. F.sub.2 =256 kHz, by means of a first low-pass filter, over a larger number of bits, then brought back to the format of the sampled signal by passing through a second sub-sampling filter, of low-pass type, over the final number of bits, 16 for example, and at the sampling frequency of F.sub.3 =16 kHz. The advantage of this device resides in the fact that the anti-aliasing filtering function is performed digitally and that the cutoff frequency is adjusted automatically as a function of the final chosen sampling frequency F.sub.3.
Such analog/digital conversion devices are very flexible in use, since it suffices to apply the signal, originating from a microphone for example, directly to the input of the analog filter, and to supply the sampling frequency F.sub.3, the frequencies F.sub.2 and F.sub.1 being able to be deduced from it.